<html><head></head><body><div style="font-family:Helvetica Neue, Helvetica, Arial, sans-serif;font-size:13px;"><div><div>Dear Joshua,</div><div><br></div><div>Thank you for the reply.</div><div><br></div><div>As you understand, I do not have SIP request to generate sip_pvt structure & a SIP channel.</div><div><br></div><div>1. Is it possible to trigger dial plan (extensions.conf - default context) and create SIP request (INVITE) on the new channel I created? Would this create a SIP channel ? </div><div> - If yes, how can I do that?</div><div> - Is it possible to bind the new channel and the SIP channel created?</div><div><br></div><div>2. What are the asterisk API's available to originate a SIP invite outside of SIP channel?</div><div><br></div><div class="ydpc1d7af59signature">Thanks, Vijay.</div></div>
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On Wednesday, November 22, 2017, 6:47:38 PM GMT+5:30, Joshua Colp <jcolp@digium.com> wrote:
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<div><div dir="ltr">On Wed, Nov 22, 2017, at 09:09 AM, vijay sukumaran nair wrote:<div class="yqt7640356344" id="yqtfd55913"><br clear="none">> Dears,<br clear="none">> I am new to asterisk. My requirement is as below, could you please review<br clear="none">> and provide me suggestions that how can I achieve the same?<br clear="none">> Requirement:1. Add a new channel, which does polling on a socket fd for<br clear="none">> an incoming message. The incoming message is proprietary protocol which<br clear="none">> commands the asterisk to perform such operation, like "CALL#1234567890".<br clear="none">> 2. On receiving the message "CALL#1234567890" the new channel should<br clear="none">> request to chan_sip to make a SIP call. Please note that the new channel<br clear="none">> which I created will not have any SIP request. <br clear="none">> What are the possible options I have, to perform #2?</div><br clear="none"><br clear="none">You can only place an outgoing to a SIP channel in that scenario, which<br clear="none">would generate a SIP INVITE and send it out. There is no interface or<br clear="none">mechanism to somehow make a SIP channel out of nowhere without any<br clear="none">signaling - that sort of defeats the entire purpose of the SIP channel<br clear="none">driver.<br clear="none"><br clear="none">-- <br clear="none">Joshua Colp<br clear="none">Digium, Inc. | Senior Software Developer<br clear="none">445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br clear="none">Check us out at: www.digium.com & www.asterisk.org<br clear="none"><br clear="none">_______________________________________________<br clear="none">asterisk-app-dev mailing list<br clear="none"><a shape="rect" ymailto="mailto:asterisk-app-dev@lists.digium.com" href="mailto:asterisk-app-dev@lists.digium.com">asterisk-app-dev@lists.digium.com</a><br clear="none"><a shape="rect" href="http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev" target="_blank">http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev</a><div class="yqt7640356344" id="yqtfd87232"><br clear="none"></div></div></div>
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